<p>Thanks Karl. I'll give the user list a try.</p>
<p>Helge</p>
<div class="gmail_quote">On May 12, 2011 5:14 PM, "Karl Schmutz" <<a href="mailto:kschmutz@starnetdata.com">kschmutz@starnetdata.com</a>> wrote:<br type="attribution">> A quick glance at your logs tells me that your SIP provider/endpoint at<br>
> <a href="http://66.8.50.218:5060">66.8.50.218:5060</a> is rejecting the call sip:**********@sip.*****.<a href="http://co.za">co.za</a><br>> SIP/2.0 <sip:**********@sip.*****.<a href="http://co.za">co.za</a>%20SIP/2.0> (line 46) by saying:<br>
> "SIP/2.0 404 Not Found"<br>> <br>> <br>> <br>> Although the number is filtered out for privacy reasons, I would double<br>> check that the dialplan at endpoint at 66.8.50.218 will accept the<br>
> number referenced in sip: ********** @ sip.*****.<a href="http://co.za">co.za</a> because that end<br>> point is actively rejecting that number.<br>> <br>> <br>> <br>> I haven't seen a post to this mailing list for a long time. You may get<br>
> better traction in the asterisk-users or in IRC as this seems to be more<br>> related to standard SIP configuration.<br>> <br>> <br>> <br>> <br>> <br>> Karl Schmutz<br>> Networking Systems Engineer<br>
> Starnet Data Design, Inc.<br>> Direct Line: 805.277.0117<br>> Toll Free: 800.779.0587<br>> Westlake Village, CA - Phoenix, AZ<br>> <a href="http://www.starnetdata.com">www.starnetdata.com</a><br>> Service. Value. Integrity.<br>
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> <br>> <br>> From: <a href="mailto:asterisk-embedded-bounces@lists.digium.com">asterisk-embedded-bounces@lists.digium.com</a><br>> [mailto:<a href="mailto:asterisk-embedded-bounces@lists.digium.com">asterisk-embedded-bounces@lists.digium.com</a>] On Behalf Of<br>
> <a href="mailto:helge.reikeras@gmail.com">helge.reikeras@gmail.com</a><br>> Sent: Thursday, May 12, 2011 7:52 AM<br>> To: <a href="mailto:asterisk-embedded@lists.digium.com">asterisk-embedded@lists.digium.com</a><br>
> Subject: [asterisk-embedded] Problem with PSTN calls (Asterisk as SIP<br>> clienton embedded device)<br>> <br>> <br>> <br>> Hi <br>> <br>> I've spent two days trying to solve this issue but to no prevail and I'm<br>
> hoping to get some help.<br>> <br>> <br>> I've configured Asterisk as a SIP client, running on OpenWRT on an<br>> embedded device with onboard FXS and ATA. Asterisk is connecting to an<br>> external SIP provider on the Internet who in turn provides a PSTN<br>
> gateway. I'm able to make calls to other SIP accounts registered on the<br>> same server who are outside my LAN. However, I can not make calls to any<br>> PSTN numbers. When trying to make PSTN calls it sounds like the person<br>
> at the other end is immediately rejecting the call although I know this<br>> is not the case.<br>> <br>> Firstly, I'm absolutely sure that the PSTN gateway is working because I<br>> can make outbound PSTN calls with the same SIP account using other SIP<br>
> clients (Empathy-SIP, SIPDroid) from the same LAN. However, when<br>> registering the same SIP account using Asterisk from OpenWRT all PSTN<br>> calls fail. Inbound calls from PSTN numbers also fail while calls from<br>
> other SIP clients on the same server work fine. The SIP accounts shows<br>> as registered in Asterisk.<br>> <br>> <br>> <br>> I've attached detailed error logs. The log files 'messages-pstn.log'<br>
> shows the failed (PSTN) call and 'messages-voip.log' shows the<br>> successful (VOIP) call. Note that I have replaced actual phone numbers<br>> and domain names with *** for anonymity.<br>> <br>> I suspect perhaps a codec issue, but I haven't been able to identify the<br>
> actual problem. Any ideas that will help me towards solving this problem<br>> is greatly appreciated.<br>> <br>> Regards,<br>> Helge<br>> <br></div>